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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: June 21 2007 at 6:24am | IP Logged
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When we wanted to send some special SIP messages using SendUdpDatagramUsingSipPort we realized that we can't have Call-ID , Branch, From-Tag and To-Tag using your API. So we should parse other messages to extract them.
Is it possible to provide us some SIP information about a Call (like above information) which may be needed when using SendUdpDatagramUsingSipPort?
Regards,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 21 2007 at 1:06pm | IP Logged
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Jalal,
Regarding access to additional SIP information - Yes we would like to do this.
Think about what you need and post a few example APIs that would do what you want. We will consider your suggestions.
We have been thinking about adding 2 new APIs to access the call's INVITE and "2xx" response. The app can then parse from these whatever it needs (tags, call ID, etc). If we give apps access to this SIP info, we would still put the onus on the app to parse out whatever it needs from the SIP messages.
We may add a SIP messaging API at a later date to make things easier but for now apps have to perform SIP message manipulations if they need to.
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: June 22 2007 at 12:48am | IP Logged
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Hi,
Our problem started when we were working on LinkSys SPA-3000 Voice Gateway. We had plugged a
Panasonic PBX Analogue Extension to the FXO port and using our Softphone connected to the
this device.
Everything was OK but we could not transfer calls on Analogue Line because we could not send
a Hook-Flash order to this device to start a new session on the analogue line.
We searched and we found we have two ways to do this.
1- RTP Telephony-Event(101) using RFC 2833
2- SIP Info message using RFC 2976
When we started to send these packets using Media Engine we encountered some new problems.
1- For the first solution we could not set appropriate parameters of RTP header unless parsing
previous RTPs sent by Media Engine. We set some of these parameters we did not know to 0.
Fortunately it worked, but I don't know why we should send 5 times of the same packets to this
device to work. Maybe it's because we did not set RTP Header parameters correctly.
2- The same problem we had with SIP INFO. Again we did not have Call-ID, Branch, Tag , ...
to make a new SIP INFO message to send to the device. But fortunately after we thiefed these
parameters from previously sent SIP packets by Media-Engine it worked perfect.
I will be happy if you share your expriences on SIP Hook-Flash orders.
I think one good idea to help your customers to have custom SIP messages would be providing
needed parameters for SIP in GetActiveCallInfo API procedure.
This function has good information now, but it can include more parameters.
Best Regards,
Jalal Abedinejad
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Get Call-ID and some more info
